The first topic when introducing someone to the world of professional audio would most certainly have to be the interface. A majority of people out there have some general notion that anyone who wants to do mixing or song creation can just grab one of the many software applications out there and give it a go. What they may not realize is: the sound card in a computer is not sufficient to the task. No matter how advanced the internal sound device may be, trying to use it for pro audio presents several logistical problems.
Noise from the fans and the processor leaches into the audio connections and while using silent running fans and shock absorbing stand-offs to cushion the motherboard may sound like a good idea, this only lessons the impact of physical vibrations and doesn’t take into account the fact that all the computer’s components are being powered by the same power supply and share a ground. This means, some of the noise being introduced is electromagnetic. While the noise is not easily heard at the beginning of a project, it becomes painfully obvious when stacking tracks and compressing.
Furthermore, the host computer is responsible for delegating resources to the various computing tasks and must, at times, reduce according to the law of supply and demand. This means that it’s impossible to guarantee a specific response time when trying to handle real-time audio. This can mean serious delays or dropouts in the tracks.
Finally, the sound card is only as functional as the software that drives it. Most drivers are written with the audio consumer, not the professional, in mind. Those that do attempt it, are usually only complying to the level of open source ASIO coding. This format will work, but it doesn’t always perform as expected. When considering this detail along with the aforementioned, it can be readily seen that no serious work should be done without an audio interface.
There are a variety of choices available on the market today. Some are very expensive and others can be purchased for as little as 200 dollars. The function of an interface is to allow you to make audio connections with your computer. They do this by replacing the sound card of the computer at least for the job at hand, and because they do their own internal processing and utilize their own power supply, this is a job they can perform with stellar results. They come with volume controls and, if being used all the time in place of the computer’s sound card, this gives you a main volume knob for the whole system. Just remember to disable the host computer’s sound device if you wish to use them as a replacement.
The next important choice to make is what DAW to use. A digital audio workstation is a software which is designed to make the work of recording, composing and mixing possible. These software kits usually come with a good selection of tools already included, but for the serious engineer, there will always be a selection of audio plugins under the hood. Because DAW software is made to host these plugins, they can certainly be added to make the software even better. Plugins come in a variety of formats, so it’s important to make sure that the selection of software matches the hardware and that the plugins will be compatible. A general bit of knowledge to point you in the right direction follows:
Mac uses Pro Tools or Logic which have their own proprietary format for plugins and
PC uses a variety of host software but can also host VST plugins.
VST is an open source format so there are many free or inexpensive plugins written in that format. It’s still a viable choice for professional work and many developers of premium plugins also create VST’s.
The last thing to consider is the fine tuning of your system. Once you’ve attached an audio interface, installed a host software and preferably added some reference monitor speakers to your setup, you should look in the audio interface’s settings for the controls over latency, bit depth and sampling rate. For pro audio, a sampling rate of 48,000 is the minimum and is equivalent to DVD audio quality. 96,000, which is a high definition, is common in pro studios; as is any rate higher than such, for instance, 192,000. This number is equivalent to the resolution on a computer’s view screen. The larger the number, the more crisp and detailed the recording and playback will be. The bit depth should be at least 24. Anywhere from 24 to 32 is common in pro studios. This number is similar to color depth and affects the way tracks mix together and respond to processing. The larger the number, the more diverse the results.
Latency is the number of milliseconds of delay there will be between the sound being generated by the computer, and the time when it’s actually heard. This number is set by you. The smaller the number is, the closer the playback will be to real-time. Most computers aren’t capable of real-time playback, even with an external audio device. They can get close enough for practical use, though. A number of 256 or lower is usually sufficient for a performer to monitor during their playing or singing. If there are skips, glitches or other noticeable artifacts, then try to dial in a number that’s a bit higher to see where the compromise lies between what the performer can take and what the system is capable of processing. If no compromise can be found because the delay required to make the playback smooth is too much of a delay for the performance, then a computer upgrade may be necessary. There are some audio interfaces, like some of those made by focusrite, that allow real-time monitoring straight from the hardware itself. If you have a slower computer, this may be the trick to give the performer what they need. Some interfaces, like the VR system made by Slate Digital and Universal Audio’s Apollo series even do some processing on the signal before it enters the computer; making advances in the modeling of specific microphones and preamps possible.
For mixing tasks, the latency should be cranked up higher. This allows for more processing power to load more plugins. Just remember, when using plugins and listening for glitches, some plugins only work at a lower sampling rate and may not be suitable for every project. I run my system at 48,000 and most plugins work fine. Anything higher requires only the best. Another tip I’ll offer here is just a suggestion. I like to set my latency at a number two notches down from the largest number when mixing. The reason I do this, is so I have a little breathing room when I inevitably get to that point in the project where the system can no longer keep up. At the point where that happens, I, then, set it to the highest number which gives the system a boost for me to finish the part I’m on before deciding how to proceed.
NOTE: IF YOU’RE BUILDING A SYSTEM FOR MULTITRACK RECORDING, THE NUMBER OF CHANNELS YOUR INTERFACE PROVIDES WILL BE KEY. HINT: YOU’LL ALWAYS NEED MORE THAN YOU THINK… AND, SOME RACK UNITS CAN BE EXPANDED. MY SAFFIRE PRO 40 UNITS CAN LINK TOGETHER AND LOOP BACK TO THE SAFFIRE OCTO PRE FOR MAXIMUM VERSATILITY.